Method and apparatus for efficient multimedia delivery in a wireless packet network

ABSTRACT

The present invention sends multiple versions of a multimedia packet to the base station, and, based on the radio channel and traffic characteristics, an appropriate version of the multimedia packet is selected to send to the mobile station at a given time. In this way, source transmission is improved to instantaneous conditions. The steps of the present invention are performed in conjunction with RTP used for multimedia transmission over internet protocol (IP) networks. In a first embodiment, the multiple versions are sent to the base station in the same RTP packet, and the base station strips out the extraneous versions. In a second embodiment, the base station receives multiple RTP packets having identical information in the packet header in many fields, and selects an appropriate one among these for transmission to the mobile station, discarding the rest.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. application Ser. No.11/933,665, filed Nov. 1, 2007, the disclosure of which is incorporatedherein by reference.

BACKGROUND

Wireless networks are used for a variety of applications. Traditionally,the main application has been voice, delivered over a circuit switchednetwork. The network has since been adapted to deliver circuit switchedand packet switched data. With higher speed networks, applications suchas video telephony will soon become available.

Within such a framework, a limited amount of functionality exists totailor the service to the environment. For example, Global System forMobile Communications (GSM) and Wideband Code Division Multiple Access(WCDMA) networks use the Adaptive Multi-Rate (AMR) vocoder for voice,which facilitates changes in the source rate with changing channelconditions in order to enable the best perceptive performance. WithinAMR, means exist to signal to a distant vocoder that receiver conditionshave changed and that a different vocoder rate is preferable.

Typically, in such a network, one vocoder is at the mobile device, andanother vocoder is at a transcoder unit or media gateway in the network.The media gateway converts the encoded voice into a form that can betransported over other networks. Thus, the rate change for voicecommunication occurs between the mobile device and the media gateway.Generally, information about the quality of the radio link is availableonly at the base station and the mobile device, and this information hasto be signaled to the vocoder, which is available locally at the mobiledevice, or is a remote node in case of the media gateway. In GSMnetworks, in-band signaling within the AMR packets is used to signal thedesired vocoder rate to the vocoder at the remote end; thus the vocoderrate can be adjusted depending on the condition of the individual radiolink. In WCDMA networks, an out-of-band signaling is used to signal thedesired vocoder rate. This is particularly applicable to circuit basedvoice transmission.

Wireless networks are gradually being converted to packet switchedarchitectures with a trend towards all application data being deliveredusing packet switching. For example, voice over IP (VoIP) is beingconsidered as the primary means of delivering voice for telephony. Mostother applications, including video, audio and other multimediaapplications will also be delivered via the packet network. Real timesource information for applications such as these is typically sentusing Real-Time Protocol (RTP) which attaches a timestamp and a sequencenumber to every packet at the source. The RTP packet is transmittedusing User Datagram Protocol (UDP) and the Internet Protocol (IP). Atthe destination, the receiver may use the timestamp information toreplay the packets at the correct relative time. Additionally, thetimestamp can be used to determine whether the packet has lost itsusefulness along the path to its destination. The sequence number isused to track lost or duplicated packets. The RTP protocol operatesbetween the source and the receiver. Some sources may send informationin a format that is not interpretable by the receiver; the media gatewaycan then convert this to a format that is understood by the receiver.Such operation is negotiated during the set up of the session. In thiscase, the RTP protocol operates between the media gateway and thereceiver.

In the IP framework, a limited amount of adaptation to the linkcapability is possible. Many sources encode data in a format that issuitable for the average quality of the link. This is often done byquerying the receiver about the link capability (e.g., 128 kbps, 1 Mbps,etc.), or can be done automatically by sounding the link and gettingfeedback as to the data capability. However, this method is not suitablefor links whose quality can vary dynamically. The quality of wirelesslinks varies at the speed of the fading, and can also vary with varyingtraffic levels. In such cases, the delay in providing feedback may belarge enough that efficient adaptation to the link is not possible. Inpacket switched radio networks, the optimum transmission format (such asvocoder rate) has to take into account the instantaneous radio channelquality, and other factors such as the traffic load and theretransmission schemes (such as hybrid ARQ). Most of this information isonly available at the base station, and not at the remote source coder.There are currently significant technical obstacles to sendinginformation on all these parameters to the remote source coder in anexpedient manner so that the coder can react to instantaneousconditions.

The so-called M-pipe project is currently being designed to overcome thesignificant technical obstacles hereinabove described. The M-pipeproject is directed to developing new media codecs that produce scalableencodings that can be locally adapted to conditions, and in thedevelopment of signaling mechanisms that can be used to convey tonetwork nodes the scalable nature of such encodings. Disadvantageously,the project lacks the ability to use existing media codecs that do notproduce scalable media.

Progressive or scalable coding is utilized in systems such as DVB-T andDVB-H, wherein hierarchical modulation schemes (e.g. 16QAM and QPSK) areused to convey source data of different fidelity to users with differentradio conditions. Such operation is mainly for broadcast transmission,and no notion of adapting to a particular user is considered in thesesystems.

In U.S. Pat. No. 7,194,000 to Balachandran and Ramesh, a method isdescribed wherein progressive encodings of source data are encoded inmultiple packets with different priorities being assigned to the packetsbased on the importance of the data. The scheduling function is capableof dropping packets of lower priority, but is still able to deliversufficient source data to the receiver in order to render the multimediainformation at some level of fidelity. Disadvantageously, this methodonly allows a coarse level of adaptation to radio conditions.

What is desired is a method and system adapted to select the best formatfor transmission without the need for such detailed feedback.

SUMMARY

The present invention provides a method by which multimedia data can bedelivered in an efficient fashion in a radio environment whereconditions can rapidly vary. The present invention further provides asystem of network architecture and network nodes with enhancedfunctionality that facilitates the implementation of the method of thepresent invention. The efficient delivery of multimedia data is achievedwith no modification to transport protocols used in IP networks (such asTCP, UDP and Internet Protocol (IP)), and minor modifications to payloadformats for RTP. In addition, the present invention presents mechanismsfor signaling functionality across a network by exploiting conventionalprotocols used in networks based on IP.

The present invention sends multiple versions of the multimedia packetto the base station, and, based on the radio channel and trafficcharacteristics, the base station selects an appropriate version to sendto the mobile station at a given time, so as to improve sourcetransmission to instantaneous conditions. This is performed inconjunction with RTP used for multimedia transmission over internetprotocol (IP) networks. In a first embodiment, the multiple versions aresent to the base station in the same RTP packet, and the base stationstrips out the extraneous versions. In a second embodiment, the basestation receives multiple RTP packets having identical information inthe packet header in many fields, and selects an appropriate one amongthese for transmission to the mobile station, discarding the rest.

BRIEF DESCRIPTION OF THE DRAWINGS

In the following section, the invention will be described with referenceto exemplary embodiments illustrated in the figures, in which:

FIG. 1 illustrates the architecture of a mobile network configured tocarry multimedia according to the present invention

FIG. 2 is a diagram of an RTP packet utilized in the implementation ofthe present invention;

FIG. 3 is a flow chart illustrating a first embodiment of the method ofthe present invention;

FIG. 4 is a flow chart illustrating a second embodiment of the method ofthe present invention;

FIG. 5 is an aspect of the present invention illustrating mediaoriginating from a transmitting mobile station;

FIG. 6 is an aspect of the present invention illustrating the originalmedia being transcoded (meaning, instead of one (normal)encoding-decoding, transcoding uses two encoding-decoding stages) in amedia gateway and then encapsulated in one or several RTP packets; and

FIG. 7 is an aspect of the present invention illustrating the severalversions of the encoded media being received and processed in a basestation and one version being sent to a receiving mobile station.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 illustrates the architecture of a mobile network 100 configuredto carry multimedia according to the present invention. As seen therein,a mobile station 101 communicates with a base station 102 via an airinterface 103. The base station 102 is connected to a backbone network104. The backbone network 104 enables communication between the mobilestation 101 and a radio access network and media gateway (not shown)then to another end terminal or server 105. In many situations, the endterminal or server 105 is capable of generating multimedia informationin formats that can be understood by the mobile station 101. In such acase, the end terminal or server 105 may also perform the functions ofencapsulating the multimedia information in an RTP packet, which may betransported using protocols such as UDP over the IP network. In othersituations, a media gateway 106 converts the signal received from theend terminal or server 105 to a format that can be understood by themobile station 101. In this case, the media gateway 106 or an associatednode may perform the encapsulation of the multimedia information withinan RTP packet.

As seen in FIG. 2, the RTP packet 200 consists of a header and payloadinformation. The header information includes a payload type field, asequence number and a timestamp. The payload type field identifies thetype of multimedia information and the multimedia codec used. Forexample, the payload type may indicate that the AMR codec was used. Thesequence number is used by the receiver to sequence packets and obtaininformation on lost packets. The timestamp gives information to themultimedia playback device at the receiver on relative times when themultimedia information needs to be played back. The specific AMR codingrate used is not part of the RTP header, but is signaled in the firstfew bits of the RTP payload. The rest of the payload includes theencoded speech bits. It will be appreciated that the payload size willchange depending on the AMR coding rate.

Referring back to FIG. 1, in many IP networks, the base station 102 (oran associated gateway 106) performs the function of protocol headercompression for many transport protocols as part of the convergencesublayer of the protocol stack. In particular, the base station 102 mayperform compression of the RTP/UDP/IP header by suppressing redundant,predictable or constant information. One such header compression scheme,robust header compression (ROHC), is defined in Internet EngineeringTask Force (IETF) Request For Comments (RFC) 3095, and is expected to becommonly used in wireless IP networks due to its robustness. It will beappreciated that the base station 102 analyzes the header of the RTPpacket to perform such compression.

In the present invention, multiple versions of the same multimediainformation are generated. On the uplink, the mobile station 101 mostlikely generates the source information, and also has access toknowledge of the channel conditions, and thus may advantageouslygenerate source information corresponding to an appropriate version. Themobile station 101, may also generate multiple AMR encoded packets forvoice at different AMR coding rates, but may only send the one mostsuited to the current conditions on the wireless channel using a singleRTP packet. Thus, the transmission of multimedia information can beimproved for the uplink in a straightforward fashion. In general, thebase station has knowledge of the channel condition experienced by themobile station's transmission in some sense. Based on such information,the base station in many wireless systems allocates transmissionresources to the mobile station. By a suitable allocation of resources,the base station may force the mobile station to use a specifictransmission format. Alternatively, the base station may also commandthe mobile station to use a specific transmission format.

On the downlink, the problem is more complicated. The multiple versions(sometimes referred to as encodings) of the same multimedia informationmay be generated at the end terminal or server 105, or can also begenerated at the media gateway 106 or in a transcoding function in thenetwork. These multiple versions of the same multimedia information maybe sent in a single RTP packet or multiple RTP packets.

FIG. 3 provides a flow chart 300 of a method of a first embodiment ofthe present invention. As seen therein, in step 301, a source or mediagateway embeds multiple source subpackets using different versions ofthe same source information (e.g., multimedia information) into the sameRTP packet, with a subheader indicating how many different versions arewithin the packet, and their lengths. In step 302 (in alternativeembodiments, step 302 may occur prior to step 301), a mobile stationfeeds back to a base station information as to its channel condition.The base station also has access to other factors such as the level ofload in the network. In step 303, the base station selects anappropriate sub-packet to send and strips out the other subpackets andthe subheader from the RTP packet before forwarding to the mobilestation in step 304, based on the information received from the mobilestation in step 302. A payload format modified from that described inIETF RFC 2198 “RTP Payload for Redundant Audio Data” may be used as anexemplary embodiment wherein multiple encodings are included in the sameRTP packet. IETF RFC 2198 describes methods to send payload data forcurrent and previous multimedia frames in order to have redundant datato protect against packet losses. However, in IETF RFC 2198, a timestampoffset is sent with the secondary data from previous frames.

In contrast, in the present invention, the same payload format as inIETF RFC 2198 can be used to send multiple payloads at the same time,but in the present invention the timestamp offset is set to zero. Apacket filter at the base station analyzes this compound packet andselects an appropriate description to send to the mobile station on thedownlink in step 303. As noted above, the base station selects anappropriate packet of the multiple versions to send to the mobilestation (after suitable header compression) depending on feedback fromthe mobile station as to its channel condition, and possibly otherfactors such as the level of load in the network. Thus, the base stationperforms the combined operation of the convergence function and thescheduling function. The scheduling operation may be chosen to improveerror performance on the link to the mobile station, or to improve thelevel of traffic being served, or an appropriate combination of the two.The present invention comprises a method for improving multimediadelivery over a wireless internet protocol (IP) network, having thesteps of generating multiple versions of a multimedia packet; sendingthe multiple versions of a multimedia packet to a base station; andbased on the operating conditions such as radio channel, network and/ortraffic characteristics, selecting an appropriate version of themultiple versions of a multimedia packet to send to a receiving mobilestation at a given time. Traffic characteristics might be localcharacteristics such as conditions affecting a stream to a specificmobile station or can be local within a specific cell that affects allusers within the cell or local in the sense of streams of the same type,such as 10 VoIP streams to 10 mobile stations within one cell. On theother hand, network characteristics have more of a global scale, forexample, a larger portion of the operators' system, such as a group ofcells, or an entire network in a region, and would include both radiochannel characteristics and backbone network characteristics, such ascongestion in routers and gateways.

FIG. 4 is a flow chart of a second embodiment 400 of the presentinvention. As seen therein, different versions of the source information(e.g., multimedia information) are generated in multiple RTP packets instep 401. The payload type, sequence number and timestamp information inthe RTP headers for these multiple packets are all the same, but thepayload is different. All these packets are sent in step 402 to the basestation using UDP/IP. Though there are multiple packets that have thesame RTP sequence number, this will be transparent to most routers andnodes in the IP network which will pass them along as routers and nodesdo not analyze packets to the level of the RTP header. It is noted thatRTP operation in conformance with IETF RFC 3550 does not explicitlypreclude the sending of multiple packets with the same sequence number,although, conventionally, sending of multiple packets with the samesequence number would not be generally expected.

In a typical wireless packet network, such as WiMAX or High Speed PacketAccess (HSPA) within WCDMA, the packet filter within the base stationanalyzes the RTP headers of the packets received within a certain timewindow appropriately chosen by the base station. If it finds multiplepackets that have the same payload type, sequence number and timestamp,but different lengths and/or different AMR rate information, then itrecognizes these packets as corresponding to multiple versions of thesame multimedia information. In step 404, the base station then selectsan appropriate packet to send to the mobile station (after suitableheader compression) depending on feedback received, in step 403, fromthe mobile station as to its channel condition, and possibly otherfactors such as the level of congestion in the network. As in the firstembodiment, the base station performs combined operation of theconvergence functions and the scheduling function. When the RTP payloadis encrypted, using protocols such as Secure Real Time Protocol (SRTP),the base station may not be able to read the AMR rate information withinthe payload and thus will be configured to make a decision based onpacket length alone. If the RTP payload is not encrypted, then thepacket filter can read the AMR rate information within, and can selectan appropriate transmission scheme at the physical layer (modulation andcoding scheme etc.) Obviously, the initiation of the packet filter inthe data path is predicated by the absence of end-to-end encryption.

In another aspect of the present invention, each AMR format can begenerated using a different contributing source, but with the sametime-stamp information, as possible within RTP because the contributingsource information can be included as part of the RTP header. In thisembodiment, the packet filter examines the incoming RTP packets havingthe same timestamp, but different contributing source information, andthus selects the appropriate source format to send over the air link.The packet filter may be part of a Media Resource Function (MRF). Atypical use of an MRF is to enable mixing of sources for conferencing.In the present invention, MRF is used to select one or more of multiplesources. In such case, the RTP packets corresponding to multiple sourcedescriptions at the same time will most likely have the same timestampbut possibly different sequence numbers, and different contributingsource numbers, which is within the strict norms of acceptableoperations as per the RTP protocol described in IETF RFC 3550. It ispossible to enable features such as conference calls in combination withthe present invention and the use of different contributing sources fordifferent descriptions of the source. In such a case, each real sourcewill have a subset of contributing source numbers associated with it forthe multiple versions. The packet filtering rule at the MRF will selectone packet with the appropriate contributing source number from eachreal source.

In another implementation of the present invention, the end terminal,server or media gateway may use another protocol header below RTP or aRTP header extension to signal to the base station that the samemultimedia information is encoded in different packets.

Further, in implementing the present invention in the case of mobile tomobile calls, a first mobile station (on the uplink) will only transmita source description appropriate for its channel conditions. In suchcase, the media gateway may perform the transcoding function andgenerate the multiple versions, which can then be sent to the MRF at thebase station. The base station will then select an appropriate versionto send to the receiving mobile station on the downlink in accordancewith the present invention. In such a case, the transcoding function atthe media gateway may be configured to only generate descriptions oflower rate than the one received from the transmitting mobile.

The present invention can also be implemented in video telephonyapplications. In this implementation, a video telephony source isconstructed that has multiple versions of the video and audio streamscorresponding to the call. Thus, the present invention enables thechoice of the video and audio description based on the channel quality,grant information or congestion. The present invention can further beextended to the case when congestion or radio channel quality precludesthe continuation of the video stream. In such case, the video stream canbe selectively suppressed, while some suitable description of the audiostream is scheduled for transmission.

The present invention can also be used in mobile station handoverscenarios. Typically, after a handover, it takes a significant amount oftime for the mobile station to learn the radio channel conditions sothat the transmission can be tailored appropriately. Using the methodsof the present invention, optimization of the transmission format to thenew radio link can commence sooner.

Referring now to FIG. 5, an aspect 500 of the present invention whereinmedia originating from a transmitting mobile station is shown. As seentherein, the original media is encoded using a plurality of encoders501A, 501B, 501C into several versions with different bit rates (packetsizes). The different versions are encapsulated in one or several RTPpackets using packetizer 502.

FIG. 6 is an aspect 600 of the present invention illustrating theoriginal media being transcoded using a plurality of transcoders 601A,601B in a media gateway and then encapsulated in one or several RTPpackets using packetizer 602. As seen therein, the originally encodedmedia is received and transcoded into several different encodingversions of the media where the different versions use different bitrates and create different packet/payload sizes. The different versionsare encapsulated in one or several RTP packets.

FIG. 7 is an aspect 700 of the present invention illustrating theseveral versions of the encoded media being received and processed in abase station 701 and one version being sent to a receiving mobilestation 702. As seen therein, several versions of the encoded media arereceived by the base station. The base station analyzes the receivedpacket(s) to find the different versions of the encoded media. Based ontraffic characteristics and/or radio channel characteristics, the basestation selects which version to send to the receiving mobile station

In each embodiment of the present invention, one function that isperformed is the instantiation of a packet filter that is selectivelycapable of processing multiple versions of sources before one or moredescriptions are transmitted over the air. The packet filter isactivated by one of several possible signaling methods:

-   -   1. The first method involves use of call control techniques such        as a SIP•INFO message or a SIP•NOTIFY message that informs the        call processing function that specialized treatment of the        source is desired before transmission over the air. The        call-control function, implemented in this example using the        Session Initiation Protocol (SIP), in turn activates a packet        processing function in the radio network. For example, the        packet processing function also known as a packet filter can be        activated for a given secondary PDP context (optionally using        the Packet Flow Identity (PFI) and the Packet Flow Context        (PFC)) at the SGSN in a GSM/EDGE Network. Alternatively, the        packet filter can be activated at a base station in a packet        network such as WiMAX by changing the classifier rule at the MAC        Layer that activates a specialized packet processing function        for specific multiple-description formals. Similar constructions        are possible in the WCDMA network to achieve the same ends.    -   2. The second method involves using the unique knowledge about        services that exists on the mobile station to request changes to        the QoS characteristics for a particular connection. In a        GSM/EDGE network or a WCDMA network, the mobile station would        send a MODIFY PDP CONTEXT message to activate specific packet        filtering functionality for a given service. In a different        wireless network such as WiMAX, the mobile station would change        the characteristics of the classifier rule at the MAC layer at        the base station to activate specialized processing functions        for specific multiple version formats.

The advantages of the present invention include as follows:

-   -   (1) Multimedia transmission is tailored to instantaneous radio        network and air interface conditions, thereby leading to a        better multimedia experience. Note that the present invention        can also be used in conjunction with schemes that allow feedback        on desired codecs to the source. Typically, such feedback is        slow, and is sent after much averaging, but can be used to        control the number of versions of multimedia information used at        the source, which can then be used by the base station to tailor        to instantaneous conditions.    -   (2) The present invention can be performed using existing        protocols used on the IP network. As a result, it has minimal        impact on existing IP network components. For example, routers        in the network will pass the IP packets of the present invention        without the need for modification in the router functions. The        UDP mechanism is also unaffected.    -   (3) No feedback is needed at the end terminal, server or media        gateway to enable such improved transmission.    -   (4) Adaptation for broadcast/multicast is enabled. For example,        a broadcast scheme in which available bandwidth (after serving        user traffic) is used to send broadcast multimedia information        can be envisioned wherein the source format that fits in the        available bandwidth can be used to support broadcast        functionality.

As will be recognized by those skilled in the art, the innovativeconcepts described in the present application can be modified and variedover a wide range of applications. Accordingly, the scope of patentedsubject matter should not be limited to any of the specific exemplaryteachings discussed above, but is instead defined by the followingclaims.

1. A method for improving multimedia delivery over a wireless internetprotocol (IP) network, comprising the steps of: generating multipleversions of a multimedia packet; sending at least one of the multipleversions of a multimedia packet to a first node in a wireless network;and selecting an appropriate version of the at least one of the multipleversions of a multimedia packet to send to a second node at a giventime.
 2. The method of claim 1, wherein the said first node is a basestation and the second node is a mobile station.
 3. The method of claim1, wherein the multiple versions of a multimedia packet are generated ata server or an end-terminal.
 4. The method of claim 1, wherein themultiple versions of a multimedia packet are generated in a transcodingfunction in a media gateway.
 5. The method of claim 1, in which a packetfilter at the first node performs the task of selecting an appropriateversion.
 6. The method of claim 1, wherein the packet filter isinstantiated using a signaling method.
 7. The method of claim 1, whereinthe multiple versions of a multimedia packet are sent in the sameReal-Time Protocol (RTP) packet.
 8. The method of claim 7, whereinsub-headers are used to identify the different versions of themultimedia packet sent in the same RTP packet.
 9. The method of claim 8,wherein the sub-headers specify timestamp offsets.
 10. The method ofclaim 9, wherein the timestamp offsets are set to zero.
 11. The methodof claim 10, wherein the first node performs header compression beforesending the chosen version of the multimedia packet.
 12. The method ofclaim 7, wherein the RTP packet format according to IETF RFC 2198 isused to send the different versions of the multimedia packet insub-packets, with the timestamp offsets between the sub-packets set tozero.
 13. The method of claim 7, wherein the first node selects theversion to send and discards other versions.
 14. The method of claim 1,wherein the multiple versions of a multimedia packet are sent indifferent RTP packets with the same sequence number and timestamp. 15.The method of claim 1, wherein the first node combines the functions ofconvergence and scheduling, wherein the scheduling operation is chosento take into account the link quality, the level of traffic beingserved, or a combination of the two.
 16. The method of claim 1, whereinthe multiple versions of a multimedia packet are sent in different RTPpackets with different contributing source information.
 17. The methodof claim 16, wherein the first node uses the contributing sourceinformation to identify different versions of the same multimediapacket, and chooses one version to send to the second node.
 18. Themethod of claim 16, for use in a multimedia conference call; whereindifferent physical sources use different sets of contributing sourceinformation values to identify multiple versions of multimedia packetscorresponding to respective physical sources.
 19. A method of enablingmobile-to-mobile multimedia calls in a wireless Internet Protocol (IP)network, comprising the steps of: sending a single version of amultimedia packet from a first mobile station to a first base station;sending the multimedia packet from the first base station to a mediagateway; generating multiple versions of the multimedia packet at themedia gateway; sending at least one of the multiple versions to a secondbase station; and selecting, by the second base station, the appropriateversion from the at least one of the multiple versions to send to asecond mobile station.
 20. A signaling method in a wireless InternetProtocol (IP) network, comprising the steps of: informing, using aSIP•INFO message, a call processing or call control function thatspecialized treatment of a source is desired before transmission overthe air, wherein the call-control function activates a packet filterfunction in the radio network.
 21. The method of claim 20, wherein theactivation of the packet filter function involves changing theclassifier rule at the MAC Layer that activates a specialized packetprocessing function for specific multiple-description formals.
 22. Aserver adapted to deliver multimedia packets over a wireless internetprotocol (IP) network, comprising: means for generating multipleversions of a multimedia packet; means for sending, to a base station,at least one of the multiple versions of the multimedia packet inindividual real time protocol (RTP) packets with the same time-stampinformation, or means for sending, to a base station, at least one ofthe multiple versions of the multimedia packet in a single real timeprotocol (RTP) packet with multiple sub-packets identified with atimestamp offset set to zero.
 23. A base station comprising: a packetfilter adapted to examine each incoming RTP packet; means foridentifying that multiple RTP packets correspond to the same multimediapacket; or means for identifying multiple versions of the multimediapacket in the same RTP packet; and means for selecting, based on theradio channel and traffic characteristics, an appropriate source formatof the RTP packet to send to a mobile station.
 24. The base station ofclaim 23, wherein the means used for identifying that multiple RTPpackets correspond to the same multimedia packet comprises means foridentifying different contributing source information for the RTPpackets, wherein a set of source information corresponding to themultimedia packets has been conveyed to the base station using asignaling procedure.
 25. A media gateway, comprising: means of receivinga single multimedia description and converting it to multipledescriptions. means to encapsulate the multiple descriptions in multipleRTP packets, wherein the RTP packets have one of the followingcharacteristics: the same timestamp and sequence number; or the sametimestamp information but different contributing source information; ormeans to assemble the multiple descriptions in a single RTP packet withsubheaders used to identify the different descriptions, wherein thesubheaders used are timestamp offsets.